Update audio code to use size_t more correctly, webrtc/modules/ portion.

webrtc/modules/utility/source/file_recorder_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 138 matching lines @@
     AudioFrame tempAudioFrame;
     tempAudioFrame.samples_per_channel_ = 0;
     if( incomingAudioFrame.num_channels_ == 2 &&
         !_moduleFile->IsStereo())
     {
         // Recording mono but incoming audio is (interleaved) stereo.
         tempAudioFrame.num_channels_ = 1;
         tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
         tempAudioFrame.samples_per_channel_ =
           incomingAudioFrame.samples_per_channel_;
-        for (uint16_t i = 0;
+        for (size_t i = 0;
              i < (incomingAudioFrame.samples_per_channel_); i++)
         {
             // Sample value is the average of left and right buffer rounded to
             // closest integer value. Note samples can be either 1 or 2 byte.
              tempAudioFrame.data_[i] =
                  ((incomingAudioFrame.data_[2 * i] +
                    incomingAudioFrame.data_[(2 * i) + 1] + 1) >> 1);
         }
     }
     else if( incomingAudioFrame.num_channels_ == 1 &&
         _moduleFile->IsStereo())
     {
         // Recording stereo but incoming audio is mono.
         tempAudioFrame.num_channels_ = 2;
         tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
         tempAudioFrame.samples_per_channel_ =
           incomingAudioFrame.samples_per_channel_;
-        for (uint16_t i = 0;
+        for (size_t i = 0;
              i < (incomingAudioFrame.samples_per_channel_); i++)
         {
             // Duplicate sample to both channels
              tempAudioFrame.data_[2*i] =
                incomingAudioFrame.data_[i];
              tempAudioFrame.data_[2*i+1] =
                incomingAudioFrame.data_[i];
         }
     }
 
@@ skipping 15 matching lines @@
     {
         if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
                                  encodedLenInBytes) == -1)
         {
             LOG(LS_WARNING) << "RecordAudioToFile() codec "
                             << codec_info_.plname
                             << " not supported or failed to encode stream.";
             return -1;
         }
     } else {
-        int outLen = 0;
+        size_t outLen = 0;
         _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
                                       codec_info_.plfreq,
                                       ptrAudioFrame->num_channels_);
         _audioResampler.Push(ptrAudioFrame->data_,
                              ptrAudioFrame->samples_per_channel_ *
                              ptrAudioFrame->num_channels_,
                              (int16_t*)_audioBuffer,
                              MAX_AUDIO_BUFFER_IN_BYTES, outLen);
         encodedLenInBytes = outLen * sizeof(int16_t);
     }
@@ skipping 35 matching lines @@
     codecInst = codec_info_;
     return 0;
 }
 
 int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
                                                 size_t bufferLength)
 {
     return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
 }
 }  // namespace webrtc